The conditions for transmitting digital information over a given channel, may be time-varying. It is therefore advantageous to have multirate data transmitters (i.e. transmitters capable of transmitting several different numbers of bits per second) provided with simple means for switching from one rate to another according to the actual transmission conditions.
The cost of the transmission lines, has induced manufacturers to look for transmission methods which allow the transmission of information from a plurality of sources, over a single channel. Digital techniques lend themselves particularly well to this type of operation. To this end, the signals originating from the various analog sources are separately sampled. These samples are digitally coded before being alternately and sequentially transmitted over a single transmission channel, using time-division multiplexing (TDM) techniques. For proper operation of the system, however, it is necessary that each intended receiver located at the other end of the transmission channel, i.e., on the receiving side, retrieves the signal intended for it. In other words, the samples must not only be properly redistributed through a demultiplexing process on the receiving side, but the information contained in the sample stream to be provided to each receiver, must be sufficient for an accurate reconstitution of the original signal. More specifically, when said sources supply speech signals to be transmitted to different called parties, the latter must receive information which is not only intelligible but also of sufficient listening quality. These are two contradictory requirement, that is, (a) efficient utilization of transmission channel bandwidth, and (b) sufficient information to enable accurate reconstruction of the original signal. On the one hand for a signal to be accurately reproduced, the digital representation of a signal must be as precise as possible, i.e., the quantity of the digital data which define this signal must be relatively large. On the other hand, the higher the quantity of data provided by a source within a given time interval, the smaller the number of the sources which can share the same transmission channel.
The solutions to this type of problem are numerous. The multiplicity of these solutions proves the interest of the industry in solving this problem and trying to find efficient solutions. Furthermore, an increased effort can at present be observed in the technical field considered, which shows that the limits to the improvements in the considered system have not yet been attained.
Two lines of research can be defined, the first one dealing with the method for processing of the signal generated by each of the sources, and the second, relating to the management of the various sources.
Since the signals considered here are essentially speech signals, voice characteristics are taken into account to define coding/decoding methods, which, for a minimum quantity of digital information, alter the quality of the speech signal, as little as possible. Many voice signal processing methods have been defined in a number of publications. More specifically, reference should be made to the book by J. Flanagan, entitled: "Speech Analysis Synthesis and Perception", published in 1965 by Springer-Verlag, to become familar with voice coding methods. Another literature source is the IEEE International Conference on Acoustics, Speech and Signal Processing" publication. There will be found therein voice compression techniques the purpose of which is to accurately code the speech signal with a minimum of digital information.
For the management of a plurality of sources for concentrating their outputs over a single transmission channel, multiplexing techniques have already been mentioned. Such techniques are roughly based on a sequential and cyclical allocation of the transmission channel to each of the sources. The limitations of such techniques can easily be understood. It is apparent that channel transmission capacity (the number of bits per second) should not, in principle, be lower than the sum of the bits supplied by the various sources during the same time. However, speech signal sources are characterized by having periods of intermittent activity. More specifically, a source which seems active at a given moment, to a person engaged in a conversation has alternating of periods of silence or inactivity embedded within periods of activity are detected. Techniques have been developed which use the presence of these periods of inactivity in order to associate a single transmission channel with a group of "L" sources, where L might seem an excessive number. TASI (Time Assignment Speech Interpolation) is one example of a channel sharing technique. Such techniques use a device for identifying the sources of the group which, at a given moment, can be considered as being effectively active (according to a predetermined criteria) and for allocating the transmission channel to these sources only. The number L is defined by statistical rules with all the risks this implies in a practical application. For instance, during periods of extensive activity, a TASI type multiplexing system may have to delay the transmission of the signals coming from some sources, or to freeze out these sources, in other words, interrupt their speech. All these solutions are, of course, unacceptable in a real-time high quality conversational system.
Finally, one can combine compression techniques which compress the speech signal supplied by each of the source, with the technique of multiplexing the group of the L sources. But there is a risk of adding an excessive complexity to the system which would jeopardize any commercial application. From amongst the methods that make use of such combinations, one can mention the method proposed by David J. Goodman in an article published in the IEEE Transactions on Communications, Vol. COM-28, No. 7, July 1980, on page 1040 and following, under the title "Embedded DPCM for Variable Bit Rate Transmission".
In Goodman, the speech signal of each of the sources of the described system is, first, coded at a relatively high bit rate (maximum rate). Next the bits obtained through coding the samples of each speech signal, are placed in the bit stream to be transmitted, according to a pre-established order based on bit weight. This order is such that, transmision of the coded signal at a transmission rate corresponding to a coding rate lower than the maximum rate, the operations to be carried out are extremely simple, i.e. bits are dropped. The number of possible transmission rates according to the techniques proposed by David J. Goodman, is however relatively limited, because only rates which are multiples of the signal sampling frequency can be used.